The present invention relates to a bandwidth compression techniques for digital audio signals using an adaptive transform coding and decoding method.
Adaptive differential pulse-code modulation (ADPCM) technique is known as a practical way of bandwidth compression and has been extensively used in digital communications. Another bandwidth compression technique that is attractive for audio frequency signals is the adaptive transform coding scheme (ATC). As described in "Adaptive Transform coding of Speech Signals", IEEE Transactions on ASSP, Vol. 25, No. 4, 1977, pages 299-309, and "Approaches to Adaptive Transform Speech Coding at Low Bit Rates", IEEE Transactions on ASSP, Vol. 27, No. 1, 1979, pages 89-95, input discrete speech samples are buffered to form a block of N speech samples each. All samples of each block are linearly transformed into a group of transform coefficients based on a linear transform. These transform coefficients are then adaptively quantized independently and transmitted. The adaptation is controlled by a short-term basis spectrum that is derived from the transform coefficients prior to quantization and transmitted as a supplementary signal to the receiver. Specifically, the short-term basis spectrum is obtained by a bit assignment process in which quantization bits are assigned corresponding to the power of the transform coefficients. At the receiver, the quantized signals are adaptively dequantized according to a supplementary signal that is derived in a manner inverse to that of the transmitter, and an inverse transform is taken to obtain the corresponding block of reconstructed speech samples.
However, in some cases where the transform coefficients exhibit a uniform distribution of power, the number of quantization levels to be assigned to each transform coefficient is smaller than is necessary to ensure a satisfactory level of signal transmission quality.